similar to: Asterisk + Cisco Call Manager

Hi, In our office, we're migrating from a Cisco set up to Asterisk. We'd like to do it gradually, so I've added an asterisk server as an H.323 gateway to the call manager so out going calls are going through asterisk. So far so good. Am now faced with the challenge relaying incoming calls from asterisk to call manager. Has anyone done that before? I won't be allowed to just make

Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands

Anyone can recommend a commercial large scale IVR with easy + pro management for asterisk? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091124/53055c0c/attachment.htm

Hello, I tried to install Asterisk + Asterisk addons + FreePBX (latest versions of all), but in the FreePBX screen, I don't have the option to set ring groups and IVRs . Can anyone tell me what I'm doing wrong? Thanks, Andreas

Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this

Thank you. I read that last yesterday afternoon, and I could've sworn I tried that but I will look into it again (I've tried so many different things it was getting cloudy what I've tried and what worked etc, combined that the extension config gets messed up after playing with it so much so I'm often recreating it as well). I also found a bug report in the FreePBX bug tracker

I?ve gotten to the bottom of this; Seems that the pjsip.endpoint_custom.conf isn?t getting included properly, or my syntax is wrong. If I put force_rport=no into pjsip.endpoint.conf and reload only Asterisk, everything works perfectly. Unfortunately, I?m using FreePBX, so it owns this file and my changes won?t persist a FreePBX reload. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F

Hi list, I've been googling this issue and found some good resources however I am still running into problems with the following combo ... Here's my story; - Asterisk 13.4 with FreePBX 12. - Migrating from Asterisk 11 / FreePBX 2.11 - Mix of Cisco 79xx handsets, mostly 7940G's. My problems started with (the very common) issue of the 7940 not replying to 401

I see that Asterisk@home and FreePBX are going along similar lines with web based management interfaces.. My Asterisk box has one analog phone, one analog line, 3 SIP phones, IAX inbound numbers, an IAX outbound trunk, IVR menus and voicemail boxes in different contexts for each of the inbound numbers.. Soon I will be adding one or more IAXy devices.. Would either Asterisk@home's or

Hello, i have Cisco Unified Communications Manager with 10 ip phone,i dont buy license IVR of Cisco Unified Communications Manager. Can i use feature IVR on Asterisk connect with Cisco Unified Communications Manager. Sorry my English.Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello users, I have a Digium Te235 and asterisk 13 which have worked well with 1 carrier but we have failed to add a 2nd carrier. The second telco brings their E1 line over finer, terminated in a RAD modem and they give me ethernet to the E1 card. It's the first time i am having install such a solution, which ideally would be not a big problem. However, The physical layer has failed to

I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware. The Asterisk setup relies heavily on queues with dynamic agents. The problem I am having is that on SOME (but not all) the Cisco phones, the phone will not allow dialing a second *. As a result, the agent can log in to queue 600 by dialing

Hi All; Using the FreePBX, after I added the extension from the GUI, I discover that it is automatically added in the extensions_additional.conf in the context [ext-local] and [from-did-direct-ivr] How I can change these context name? I need to determine this. How? Regards Bilal

On Tue, Jun 30, 2015 at 3:34 AM, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13 which have worked well with 1 >> carrier but

What response do you get to *CLI> pri show spans ? On 30 June 2015 at 09:34, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13

Hi, I'm actually using a slightly old version of AAH with Asterisk 1.2.1, because at first install it was perfect for my moderate knowledge of Asterisk. It is working well but I gradually introduced many changes to dialplan during normal use and now I'm feeling like in a straitjacket! Moreover I'd like to have the chance to upgrade Asterisk regularly. I have not the experience to

Steve Edwards wrote: > 0) I hope you mean you want to run Asterisk at home instead of > 'Asterisk at Home.' A at H was an ancient distribution from around 2005. Yes of course I didn't mean an ancient distro from 2005. > > 1) Rent a DID (a 'PSTN number') from a reputable SIP provider. This > eliminates the need for a PCI/USB interface and you won't disrupt

Hi, We currently testing a trixbox/asterisk installation and have used Freepbx to set-up and configure the box and it is running tremendously well. We have an generic IVR configured to which can transfer callers to a child IVR. This child IVR has a number of options to send the caller off to various queues. However we would like to protect some of the options with a password/pin number so that

Hello; I certainly appreciate your response. In fact, I used that exact solution for three of the incoming lines. I setup ring groups and a silent ringtone for each phone. Unfortunately, the last incoming line is more complicated and uses an IVR with multiple input choices, so the solution is not as clear cut as for the other ones. That's why I was trying to look at other options. Best

Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I've tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file.